[asterisk-commits] branch oej/moduletest r12543 - in /team/oej/moduletest: ./ channels/chan_sip.c

[asterisk-commits at lists.digium.com]

Author: oej
Date: Sun Mar 12 10:28:45 2006
New Revision: 12543

URL: http://svn.digium.com/view/asterisk?rev=12543&view=rev
Log:
Resolve conflict, reset automerge

Modified:
    team/oej/moduletest/   (props changed)
    team/oej/moduletest/channels/chan_sip.c

Propchange: team/oej/moduletest/
------------------------------------------------------------------------------
    automerge = http://edvina.net/training

Propchange: team/oej/moduletest/
------------------------------------------------------------------------------
--- svnmerge-integrated (original)
+++ svnmerge-integrated Sun Mar 12 10:28:45 2006
@@ -1,1 +1,1 @@
-/branches/1.2:1-7496,7498-12460
+/branches/1.2:1-7496,7498-12539

Modified: team/oej/moduletest/channels/chan_sip.c
URL: http://svn.digium.com/view/asterisk/team/oej/moduletest/channels/chan_sip.c?rev=12543&r1=12542&r2=12543&view=diff
==============================================================================
--- team/oej/moduletest/channels/chan_sip.c (original)
+++ team/oej/moduletest/channels/chan_sip.c Sun Mar 12 10:28:45 2006
@@ -2451,13 +2451,15 @@
 	if (!ast_test_flag(p, SIP_ALREADYGONE) && !ast_strlen_zero(p->initreq.data)) {
 		if (needcancel) {	/* Outgoing call, not up */
 			if (ast_test_flag(p, SIP_OUTGOING)) {
+				/* stop retransmitting an INVITE that has not received a response */
+				__sip_pretend_ack(p);
+
+				/* Send a new request: CANCEL */
 				transmit_request_with_auth(p, SIP_CANCEL, p->ocseq, 1, 0);
 				/* Actually don't destroy us yet, wait for the 487 on our original 
 				   INVITE, but do set an autodestruct just in case we never get it. */
 				ast_clear_flag(&locflags, SIP_NEEDDESTROY);
-				sip_scheddestroy(p, 15000);
-				/* stop retransmitting an INVITE that has not received a response */
-				__sip_pretend_ack(p);
+				sip_scheddestroy(p, 32000);
 				if ( p->initid != -1 ) {
 					/* channel still up - reverse dec of inUse counter
 					   only if the channel is not auto-congested */
@@ -2487,12 +2489,34 @@
 	return 0;
 }
 
+/*! \brief Try setting codec suggested by the SIP_CODEC channel variable */
+static void try_suggested_sip_codec(struct sip_pvt *p)
+{
+	int fmt;
+	char *codec;
+
+	codec = pbx_builtin_getvar_helper(p->owner, "SIP_CODEC");
+	if (!codec) 
+		return;
+
+	fmt = ast_getformatbyname(codec);
+	if (fmt) {
+		ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
+		if (p->jointcapability & fmt) {
+			p->jointcapability &= fmt;
+			p->capability &= fmt;
+		} else
+			ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
+	} else
+		ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
+	return;	
+}
+
 /*! \brief  sip_answer: Answer SIP call , send 200 OK on Invite 
  * Part of PBX interface */
 static int sip_answer(struct ast_channel *ast)
 {
-	int res = 0,fmt;
-	const char *codec;
+	int res = 0;
 	struct sip_pvt *p = ast->tech_pvt;
 
 	ast_mutex_lock(&p->lock);
@@ -2500,19 +2524,7 @@
 #ifdef OSP_SUPPORT	
 		time(&p->ospstart);
 #endif
-	
-		codec=pbx_builtin_getvar_helper(p->owner,"SIP_CODEC");
-		if (codec) {
-			fmt=ast_getformatbyname(codec);
-			if (fmt) {
-				ast_log(LOG_NOTICE, "Changing codec to '%s' for this call because of ${SIP_CODEC) variable\n",codec);
-				if (p->jointcapability & fmt) {
-					p->jointcapability &= fmt;
-					p->capability &= fmt;
-				} else
-					ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because it is not shared by both ends.\n");
-			} else ast_log(LOG_NOTICE, "Ignoring ${SIP_CODEC} variable because of unrecognized/not configured codec (check allow/disallow in sip.conf): %s\n",codec);
-		}
+		try_suggested_sip_codec(p);	
 
 		ast_setstate(ast, AST_STATE_UP);
 		if (option_debug)
@@ -4517,6 +4529,7 @@
 	}
 	respprep(&resp, p, msg, req);
 	if (p->rtp) {
+		try_suggested_sip_codec(p);	
 		add_sdp(&resp, p);
 	} else {
 		ast_log(LOG_ERROR, "Can't add SDP to response, since we have no RTP session allocated. Call-ID %s\n", p->callid);